Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
[asterisk/asterisk.git] / res / pjproject / tests / pjsua / scripts-sendto / 320_srtp_with_unknown_media_2.py
1 # $Id$
2 import inc_sip as sip
3 import inc_sdp as sdp
4
5 sdp = \
6 """
7 v=0
8 o=- 0 0 IN IP4 127.0.0.1
9 s=-
10 c=IN IP4 127.0.0.1
11 t=0 0
12 m=xapplicationx 4000 RTP/AVP 100
13 a=rtpmap:100 myapp/80000
14 m=audio 5000 RTP/AVP 0
15 a=crypto:1 aes_cm_128_hmac_sha1_80 inline:WnD7c1ksDGs+dIefCEo8omPg4uO8DYIinNGL5yxQ
16 """
17
18 pjsua_args = "--null-audio --auto-answer 200 --use-srtp 1 --srtp-secure 0"
19 extra_headers = ""
20 include = ["Content-Type: application/sdp",     # response must include SDP
21            "m=xapplicationx 0 RTP/AVP[\\s\\S]+m=audio [1-9]+[0-9]* RTP/AVP[\\s\\S]+a=crypto"
22            ]
23 exclude = []
24
25 sendto_cfg = sip.SendtoCfg("Unknown media and SRTP audio", pjsua_args, sdp, 200,
26                            extra_headers=extra_headers,
27                            resp_inc=include, resp_exc=exclude) 
28